mirror of https://bitbucket.org/ausocean/av.git
revid: added audio.go to handle sound cards and devices
audio.go will be used for recording sound from the sound card and mic it is like audio-netsender but it is a package instead of a command and without the netsender.
This commit is contained in:
Trek H
parent
e0ec6a8dc9
commit
58b9458ff4
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@ -0,0 +1,555 @@
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/*
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NAME
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audio-netsender - NetSender client for sending audio to NetReceiver
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AUTHORS
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Alan Noble <alan@ausocean.org>
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Trek Hopton <trek@ausocean.org>
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ACKNOWLEDGEMENTS
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A special thanks to Joel Jensen for his Go ALSA package.
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LICENSE
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audio-netsender is Copyright (C) 2018 the Australian Ocean Lab (AusOcean).
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It is free software: you can redistribute it and/or modify them under
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the terms of the GNU General Public License as published by the
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Free Software Foundation, either version 3 of the License, or (at your
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option) any later version.
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It is distributed in the hope that it will be useful, but WITHOUT
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ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
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FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
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for more details.
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You should have received a copy of the GNU General Public License
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along with https://bitbucket.org/ausocean/iot/src/master/gpl.txt.
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If not, see http://www.gnu.org/licenses.
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*/
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// audio-netsender is a NetSender client for sending audio to
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// NetReceiver. Audio is captured by means of an ALSA recording
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// device, specified by the NetReceiver "source" variable. It sent via
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// HTTP to NetReceiver in raw audio form, i.e., as PCM data, where it
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// is stored as BinaryData objects. Other NetReceiver variables are
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// "rate", "period", "channels" and "bits", for specifiying the frame
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// rate (Hz), audio period (seconds), number of channels and sample
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// bit size respectively. For a description of NetReceiver see
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// http://netreceiver.appspot.com/help.
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package main
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import (
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"errors"
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"flag"
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"io"
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"strconv"
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"sync"
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"time"
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"github.com/yobert/alsa"
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"bitbucket.org/ausocean/av/codec/pcm"
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"bitbucket.org/ausocean/iot/pi/netsender"
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"bitbucket.org/ausocean/iot/pi/sds"
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"bitbucket.org/ausocean/iot/pi/smartlogger"
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"bitbucket.org/ausocean/utils/logger"
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"bitbucket.org/ausocean/utils/ring"
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)
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const (
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progName = "audio-netsender"
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logPath = "/var/log/netsender"
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retryPeriod = 5 * time.Second
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defaultFrameRate = 48000
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defaultPeriod = 5 // seconds
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defaultChannels = 2
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defaultBits = 16
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rbDuration = 300 // seconds
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rbTimeout = 100 * time.Millisecond
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rbNextTimeout = 100 * time.Millisecond
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)
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// audioClient holds everything we need to know about the client.
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// NB: At 44100 Hz frame rate, 2 channels and 16-bit samples, a period of 5 seconds
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// results in PCM data chunks of 882000 bytes! A longer period exceeds datastore's 1MB blob limit.
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type audioClient struct {
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mu sync.Mutex // mu protects the audioClient.
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parameters
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// internals
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dev *alsa.Device // audio input device
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ab alsa.Buffer // ALSA's buffer
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rb *ring.Buffer // our buffer
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ns *netsender.Sender // our NetSender
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vs int // our "var sum" to track var changes
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}
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type parameters struct {
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mode string // operating mode, either "Normal" or "Paused"
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source string // name of audio source, or empty for the default source
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rate int // frame rate in Hz, 44100Hz by default
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period int // audio period in seconds, 5s by default
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channels int // number of audio channels, 1 for mono, 2 for stereo
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bits int // sample bit size, 16 by default
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}
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var log *logger.Logger
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func main() {
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var logLevel int
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flag.IntVar(&logLevel, "LogLevel", int(logger.Debug), "Specifies log level")
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flag.Parse()
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validLogLevel := true
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if logLevel < int(logger.Debug) || logLevel > int(logger.Fatal) {
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logLevel = int(logger.Info)
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validLogLevel = false
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}
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logSender := smartlogger.New(logPath)
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log = logger.New(int8(logLevel), &logSender.LogRoller)
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log.Log(logger.Info, "log-netsender: Logger Initialized")
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if !validLogLevel {
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log.Log(logger.Error, "Invalid log level was defaulted to Info")
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}
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var ac audioClient
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var err error
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ac.ns, err = netsender.New(log, nil, sds.ReadSystem, nil)
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if err != nil {
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log.Log(logger.Fatal, "netsender.Init failed", "error", err.Error())
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}
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// Get audio params and store the current var sum.
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vars, err := ac.ns.Vars()
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if err != nil {
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log.Log(logger.Warning, "netsender.Vars failed; using defaults", "error", err.Error())
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}
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ac.params(vars)
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ac.vs = ac.ns.VarSum()
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// Open the requested audio device.
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err = ac.open()
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if err != nil {
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log.Log(logger.Fatal, "alsa.open failed", "error", err.Error())
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}
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// Capture audio in periods of ac.period seconds, and buffer rbDuration seconds in total.
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ac.ab = ac.dev.NewBufferDuration(time.Second * time.Duration(ac.period))
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recSize := (((len(ac.ab.Data) / ac.dev.BufferFormat().Channels) * ac.channels) / ac.dev.BufferFormat().Rate) * ac.rate
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rbLen := rbDuration / ac.period
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ac.rb = ring.NewBuffer(rbLen, recSize, rbTimeout)
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go ac.input()
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ac.output()
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}
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// params extracts audio params from corresponding NetReceiver vars and returns true if anything has changed.
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// See audioClient for a description of the params and their limits.
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func (ac *audioClient) params(vars map[string]string) bool {
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// We are the only writers to this field
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// so we don't need to lock here.
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p := ac.parameters
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changed := false
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mode := vars["mode"]
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if p.mode != mode {
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p.mode = mode
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changed = true
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}
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source := vars["source"]
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if p.source != source {
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p.source = source
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changed = true
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}
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val, err := strconv.Atoi(vars["rate"])
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if err != nil {
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val = defaultFrameRate
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}
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if p.rate != val {
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p.rate = val
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changed = true
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}
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val, err = strconv.Atoi(vars["period"])
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if err != nil || val < 1 || 5 < val {
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val = defaultPeriod
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}
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if p.period != val {
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p.period = val
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changed = true
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}
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val, err = strconv.Atoi(vars["channels"])
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if err != nil || (val != 1 && val != 2) {
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val = defaultChannels
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}
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if p.channels != val {
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p.channels = val
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changed = true
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}
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val, err = strconv.Atoi(vars["bits"])
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if err != nil || (val != 16 && val != 32) {
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val = defaultBits
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}
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if p.bits != val {
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p.bits = val
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changed = true
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}
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if changed {
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ac.mu.Lock()
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ac.parameters = p
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ac.mu.Unlock()
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log.Log(logger.Debug, "Params changed")
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}
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log.Log(logger.Debug, "Parameters", "mode", p.mode, "source", p.source, "rate", p.rate, "period", p.period, "channels", p.channels, "bits", p.bits)
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return changed
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}
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// open or re-open the recording device with the given name and prepare it to record.
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// If name is empty, the first recording device is used.
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func (ac *audioClient) open() error {
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if ac.dev != nil {
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log.Log(logger.Debug, "Closing", "source", ac.source)
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ac.dev.Close()
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ac.dev = nil
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}
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log.Log(logger.Debug, "Opening", "source", ac.source)
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cards, err := alsa.OpenCards()
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if err != nil {
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return err
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}
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defer alsa.CloseCards(cards)
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for _, card := range cards {
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devices, err := card.Devices()
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if err != nil {
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return err
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}
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for _, dev := range devices {
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if dev.Type != alsa.PCM || !dev.Record {
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continue
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}
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if dev.Title == ac.source || ac.source == "" {
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ac.dev = dev
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break
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}
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}
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}
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if ac.dev == nil {
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return errors.New("No audio source found")
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}
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log.Log(logger.Debug, "Found audio source", "source", ac.dev.Title)
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// ToDo: time out if Open takes too long.
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err = ac.dev.Open()
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if err != nil {
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return err
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}
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log.Log(logger.Debug, "Opened audio source")
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_, err = ac.dev.NegotiateChannels(defaultChannels)
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if err != nil {
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return err
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}
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// Try to negotiate a rate to record in that is divisible by the wanted rate
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// so that it can be easily downsampled to the wanted rate.
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// Note: if a card thinks it can record at a rate but can't actually, this can cause a failure. Eg.
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// the audioinjector is supposed to record at 8000Hz and 16000Hz but it can't due to a firmware issue,
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// to fix this 8000 and 16000 must be removed from this slice.
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rates := [8]int{8000, 16000, 32000, 44100, 48000, 88200, 96000, 192000}
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foundRate := false
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for i := 0; i < len(rates) && !foundRate; i++ {
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if rates[i] < ac.rate {
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continue
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}
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if rates[i]%ac.rate == 0 {
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_, err = ac.dev.NegotiateRate(rates[i])
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if err == nil {
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foundRate = true
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log.Log(logger.Debug, "Sample rate set", "rate", rates[i])
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}
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}
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}
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// If no easily divisible rate is found, then use the default rate.
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if !foundRate {
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log.Log(logger.Warning, "No available device sample-rates are divisible by the requested rate. Default rate will be used. Resampling may fail.", "rateRequested", ac.rate)
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_, err = ac.dev.NegotiateRate(defaultFrameRate)
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if err != nil {
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return err
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}
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log.Log(logger.Debug, "Sample rate set", "rate", defaultFrameRate)
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}
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var fmt alsa.FormatType
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switch ac.bits {
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case 16:
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fmt = alsa.S16_LE
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case 32:
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fmt = alsa.S32_LE
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default:
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return errors.New("Unsupported sample bits")
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}
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_, err = ac.dev.NegotiateFormat(fmt)
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if err != nil {
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return err
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}
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// Either 8192 or 16384 bytes is a reasonable ALSA buffer size.
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_, err = ac.dev.NegotiateBufferSize(8192, 16384)
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if err != nil {
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return err
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}
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if err = ac.dev.Prepare(); err != nil {
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return err
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}
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log.Log(logger.Debug, "Successfully negotiated ALSA params")
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return nil
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}
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// input continously records audio and writes it to the ringbuffer.
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// Re-opens the device and tries again if ASLA returns an error.
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// Spends a lot of time sleeping in Paused mode.
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// ToDo: Currently, reading audio and writing to the ringbuffer are synchronous.
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// Need a way to asynchronously read from the ALSA buffer, i.e., _while_ it is recording to avoid any gaps.
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func (ac *audioClient) input() {
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for {
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ac.mu.Lock()
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mode := ac.mode
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ac.mu.Unlock()
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if mode == "Paused" {
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time.Sleep(time.Duration(ac.period) * time.Second)
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continue
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}
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log.Log(logger.Debug, "Recording audio for period", "seconds", ac.period)
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ac.mu.Lock()
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err := ac.dev.Read(ac.ab.Data)
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ac.mu.Unlock()
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if err != nil {
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log.Log(logger.Debug, "Device.Read failed", "error", err.Error())
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ac.mu.Lock()
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err = ac.open() // re-open
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if err != nil {
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log.Log(logger.Fatal, "alsa.open failed", "error", err.Error())
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}
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ac.mu.Unlock()
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continue
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}
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toWrite := ac.formatBuffer()
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log.Log(logger.Debug, "Audio format conversion has been performed where needed")
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var n int
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n, err = ac.rb.Write(toWrite.Data)
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switch err {
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case nil:
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log.Log(logger.Debug, "Wrote audio to ringbuffer", "length", n)
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case ring.ErrDropped:
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log.Log(logger.Warning, "Dropped audio")
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default:
|
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log.Log(logger.Error, "Unexpected ringbuffer error", "error", err.Error())
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return
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}
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}
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}
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// output continously reads audio from the ringbuffer and sends it to NetReceiver via poll requests.
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// When "B0" is configured as one of the NetReceiver inputs, audio data is posted as "B0".
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// When "B0" is not an input, the poll request happens without any audio data
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// (although other inputs may still be present via URL parameters).
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// When paused, polling continues but without sending audio (B0) data.
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// Sending is throttled so as to complete one pass of this loop approximately every audio period,
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// since cycling more frequently is pointless.
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// Finally while audio data is sent every audio period, other data is reported only every monitor period.
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// This function also handles NetReceiver configuration requests and updating of NetReceiver vars.
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func (ac *audioClient) output() {
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// Calculate the size of the output data based on wanted channels and rate.
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outLen := (((len(ac.ab.Data) / ac.ab.Format.Channels) * ac.channels) / ac.ab.Format.Rate) * ac.rate
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buf := make([]byte, outLen)
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|
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mime := "audio/x-wav;codec=pcm;rate=" + strconv.Itoa(ac.rate) + ";channels=" + strconv.Itoa(ac.channels) + ";bits=" + strconv.Itoa(ac.bits)
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ip := ac.ns.Param("ip")
|
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mp, err := strconv.Atoi(ac.ns.Param("mp"))
|
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if err != nil {
|
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log.Log(logger.Fatal, "mp not an integer")
|
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}
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|
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report := true // Report non-audio data.
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reported := time.Now() // When we last did so.
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for {
|
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var reconfig bool
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start := time.Now()
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audio := false
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var pins []netsender.Pin
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if ac.mode == "Paused" {
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// Only send X data when paused (if any).
|
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if report {
|
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pins = netsender.MakePins(ip, "X")
|
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}
|
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} else {
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n, err := read(ac.rb, buf)
|
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if err != nil {
|
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return
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}
|
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if n == 0 {
|
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goto sleep
|
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}
|
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if n != len(buf) {
|
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log.Log(logger.Error, "Unexpected length from read", "length", n)
|
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return
|
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}
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if report {
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pins = netsender.MakePins(ip, "")
|
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} else {
|
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pins = netsender.MakePins(ip, "B")
|
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}
|
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for i, pin := range pins {
|
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if pin.Name == "B0" {
|
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audio = true
|
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pins[i].Value = n
|
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pins[i].Data = buf
|
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pins[i].MimeType = mime
|
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}
|
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}
|
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}
|
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|
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if !(report || audio) {
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goto sleep // nothing to do
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}
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|
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// Populate X pins, if any.
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for i, pin := range pins {
|
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if pin.Name[0] == 'X' {
|
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err := sds.ReadSystem(&pins[i])
|
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if err != nil {
|
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log.Log(logger.Warning, "sds.ReadSystem failed", "error", err.Error())
|
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// Pin.Value defaults to -1 upon error, so OK to continue.
|
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}
|
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}
|
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}
|
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_, reconfig, err = ac.ns.Send(netsender.RequestPoll, pins)
|
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if err != nil {
|
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log.Log(logger.Debug, "netsender.Send failed", "error", err.Error())
|
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goto sleep
|
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}
|
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if report {
|
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reported = start
|
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report = false
|
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}
|
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if reconfig {
|
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err = ac.ns.Config()
|
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if err != nil {
|
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log.Log(logger.Warning, "netsender.Config failed", "error", err.Error())
|
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goto sleep
|
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}
|
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ip = ac.ns.Param("ip")
|
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mp, err = strconv.Atoi(ac.ns.Param("mp"))
|
||||
if err != nil {
|
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log.Log(logger.Fatal, "mp not an integer")
|
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}
|
||||
}
|
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|
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if ac.vs != ac.ns.VarSum() {
|
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vars, err := ac.ns.Vars()
|
||||
if err != nil {
|
||||
log.Log(logger.Error, "netsender.Vars failed", "error", err.Error())
|
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goto sleep
|
||||
}
|
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ac.params(vars) // ToDo: re-open device if audio params have changed.
|
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ac.vs = ac.ns.VarSum()
|
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}
|
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|
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sleep:
|
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pause := ac.period*1000 - int(time.Since(start).Seconds()*1000)
|
||||
if pause > 0 {
|
||||
time.Sleep(time.Duration(pause) * time.Millisecond)
|
||||
}
|
||||
if time.Since(reported).Seconds() >= float64(mp) {
|
||||
report = true
|
||||
}
|
||||
|
||||
}
|
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}
|
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|
||||
// read reads a full PCM chunk from the ringbuffer, returning the number of bytes read upon success.
|
||||
// Any errors returned are unexpected and should be considered fatal.
|
||||
func read(rb *ring.Buffer, buf []byte) (int, error) {
|
||||
chunk, err := rb.Next(rbNextTimeout)
|
||||
switch err {
|
||||
case nil:
|
||||
// Do nothing.
|
||||
case ring.ErrTimeout:
|
||||
return 0, nil
|
||||
case io.EOF:
|
||||
log.Log(logger.Error, "Unexpected EOF from ring.Next")
|
||||
return 0, io.ErrUnexpectedEOF
|
||||
default:
|
||||
log.Log(logger.Error, "Unexpected error from ring.Next", "error", err.Error())
|
||||
return 0, err
|
||||
}
|
||||
|
||||
n, err := io.ReadFull(rb, buf[:chunk.Len()])
|
||||
if err != nil {
|
||||
log.Log(logger.Error, "Unexpected error from ring.Read", "error", err.Error())
|
||||
return n, err
|
||||
}
|
||||
|
||||
log.Log(logger.Debug, "Read audio from ringbuffer", "length", n)
|
||||
return n, nil
|
||||
}
|
||||
|
||||
// formatBuffer returns an ALSA buffer that has the recording data from the ac's original ALSA buffer but stored
|
||||
// in the desired format specified by the ac's parameters.
|
||||
func (ac *audioClient) formatBuffer() alsa.Buffer {
|
||||
var err error
|
||||
ac.mu.Lock()
|
||||
wantChannels := ac.channels
|
||||
wantRate := ac.rate
|
||||
ac.mu.Unlock()
|
||||
|
||||
// If nothing needs to be changed, return the original.
|
||||
if ac.ab.Format.Channels == wantChannels && ac.ab.Format.Rate == wantRate {
|
||||
return ac.ab
|
||||
}
|
||||
|
||||
formatted := alsa.Buffer{Format: ac.ab.Format}
|
||||
bufCopied := false
|
||||
if ac.ab.Format.Channels != wantChannels {
|
||||
|
||||
// Convert channels.
|
||||
if ac.ab.Format.Channels == 2 && wantChannels == 1 {
|
||||
if formatted.Data, err = pcm.StereoToMono(ac.ab); err != nil {
|
||||
log.Log(logger.Warning, "Channel conversion failed, audio has remained stereo", "error", err.Error())
|
||||
} else {
|
||||
formatted.Format.Channels = 1
|
||||
}
|
||||
bufCopied = true
|
||||
}
|
||||
}
|
||||
|
||||
if ac.ab.Format.Rate != wantRate {
|
||||
|
||||
// Convert rate.
|
||||
if bufCopied {
|
||||
formatted.Data, err = pcm.Resample(formatted, wantRate)
|
||||
} else {
|
||||
formatted.Data, err = pcm.Resample(ac.ab, wantRate)
|
||||
}
|
||||
if err != nil {
|
||||
log.Log(logger.Warning, "Rate conversion failed, audio has remained original rate", "error", err.Error())
|
||||
} else {
|
||||
formatted.Format.Rate = wantRate
|
||||
}
|
||||
}
|
||||
return formatted
|
||||
}
|
||||
Loading…
Reference in New Issue